• Journal of Broadcast Engineering
• /
• v.13 no.1
• /
• pp.34-42
• /
• 2008

This paper proposes the mechanism which improves the qualify of video on a limited network bandwidth by applying the ROI technique to an H.264 Scalable Extension technique. The network environment assumed in this parer is the next generation network convergence environment in which the mobile device has one or more network interfaces. Therefore, we allocate the priority to video packets as the hierarchy structure of H.264 SVC-encoded video stream and ROI information, and transmit those packets over appropriate network channel for using those multiple network interfaces. This paper shows two experiments first one is extracting and allocating the video stream on an appropriate network channel, second one is unequal packet transmission by allocated priorities (e.g. ROI). Performance evaluations show that this approach delivers an improved decoded video quality when compared with conventional transmission schemes, especially on device which has multiple network interfaces.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.12B
• /
• pp.1183-1191
• /
• 2010

In this paper, I propose an adaptive OFDM CP length algorithm for in PLC systems for smart grid. The proposed scheme calculates the channel delay information at the CP controller of the receiver by taking correlation between a received data frame and the following delayed one. The CP controller, immediately, feeds back the channel delay information to the transmitter. Then, the transmitter adapts CP length for next data frame. As an impulsive noise model, Middleton Class A interference model was employed. The performance is evaluated in terms of packet data rate, cumulative packet data rate, and bit error rate (BER). The simulation results showed data gain (which is the amount of the reduced bits) gets larger as the number of packets increase, but the amount of data gain reduced as the number of branches ($N_{br}$) increase. In respects of BER for the cases $N_{br}$ is 3, 4, and 5, performance of the adaptive CP length algorithm and the fixed CP scheme are similar. Therefore, it is confirmed the proposed scheme achieved data rate increment without BER performance reduction compared to the conventional fixed CP length scheme.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.3B
• /
• pp.380-389
• /
• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.10 s.352
• /
• pp.79-89
• /
• 2006

TCP is the most widely used transport protocol in Internet applications that guarantees a reliable data transfer. But, in the wireless multi-hop networks, TCP performance is degraded because it is designed for wired networks. The main reasons of TCP performance degradation are contention for wireless medium at the MAC layer, hidden terminal problem, exposed terminal problem, packet losses in the link layer, unfairness problem, reordering problem caused by path disconnection, bandwidth waste caused by exponential backoff of retransmission timer due to node's mobility and so on. Specially, in the mobile ad-hoc networks, discrepancy between a station's transmission range and interference range produces hidden terminal problem that decreases TCP performance greatly by limiting simultaneous transmission at a time. In this paper, we propose a new MAC algorithm for mobile ad-hoc networks to solve the problem that a node can not transmit and just increase CW by hidden terminal. In the IEEE 802.11 MAC DCF, a node increases CW exponentially when it fails to transmit, but the proposed algorithm, changes CW adaptively according to the reason of failure so we get a TCP performance enhancement. We show by ns-2 simulation that the proposed algorithm enhances the TCP performance by fairly distributing the transmission opportunity to the failed nodes by hidden terminal problems.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.12 no.6
• /
• pp.229-240
• /
• 2012

The development of multimedia applications followed by development of high-speed networks has improved the performance of mobile devices with high transfer speed broadband. Mobile internet access has made possible seamless indoor and outdoor mobile multicast services. Multicasting services are used to support efficient group communications. However, mobile multicasting services have two constraints: tunnel convergence and handover latency. Many protocols and handover methods have been proposed to address these problems. The inter-LMA optimized handover model for multicasting services has previously been proposed for PMIPv6-based networks. The proposed model removes the tunnel convergence issue and reduces router processing costs. It also makes possible the performance of fast handover operations with adaptive transmission mechanisms. In addition, the proposed scheme exhibits low packet delivery costs and handover latency in comparison with existing schemes, and ensures fast handover when moving the inter-LMA domain

• Journal of the Institute of Electronics and Information Engineers
• /
• v.49 no.11
• /
• pp.132-138
• /
• 2012

VANET(Vehicular Ad-hoc Network), standardization of IEEE 802.11p specification is in process. 802.11 MAC protocol grants all nodes equal opportunity to acquire channel without regard to their bit-rates, making it possible for lower bit-rate nodes to occupy communication channel for a fair amount of time thus keeping the higher bit-rate nodes from acquiring connection channel which downward-equalize the overall network performance. Also with the 802.11p MAC protocol, the probability of collision occurring increases as the number of nodes grow. The proposed algorithm is a new MAC protocol that guarantees nodes with acquired channel a firm priority over other nodes for a fixed amount of time with TXOP concept added to 'packet burst' according to the current transmitting speed. This newly designed algorithm allows the construction of wireless network with enhanced network throughput, decreased probability of collisions as well as providing the means to grant each node a fair chance of acquiring connection according to their channel conditions. The algorithm sets the CW's (Contention Window) width wider than the standard's and modulates the continuous transmitting threshold value depending on channel acquired time, thus improving the overall performance of the network.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.40 no.8
• /
• pp.1613-1621
• /
• 2015

This paper analyzes the rate adaptation scheme and suggests applicable strategy of the MCS(Modulation and Coding Scheme) for improving DCF throughput in the IEEE 802.11ad and 802.11ad wireless LAN. IEEE 802.11ac and 802.11ad wireless LAN provide MCS technique that dynamically adjusts modulation level and code rate to the time-varying channel conditions in order to obtain considerably high data rates. But these standards did not provide rate adaptation algorithm, so this paper surveyes rate adaptation algorithm and suggests MCS scheme applied to IEEE 802.11ac and 802.11ad wireless LAN. Specially A MAC(Medium Access Control) layer throughput is evaluated over error-prone channel in the IEEE 802.11ac-based wireless LAN. In this evaluation, DCF (Distributed Coordination Function) protocol and A-MPDU (MAC Protocol Data Unit Aggregation) scheme are used. Using theoretical analysis method, the MAC saturation throughput is evaluated with the PER (Packet Error Rate) on the condition that the number of station, transmission probability, the number of parallel beams and the number of frames in each A-MPDU are variables.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.41 no.7
• /
• pp.1-8
• /
• 2004

A simple AMC (Adaptive Modulation and Coding) technique using ARQ (Automatic Repeat Request) for a MIMO (Multiple Input Multiple Output) system is proposed which does not require the additional feedback. In addition, the proposed AMC technique is different from the conventional technique in the aspect of considering the MCS (Modulation and Coding Scheme) level from the previous packet. The proposed technique can discard fewer amounts of unsuitable packets than the conventional technique. In the proposed system not only same rate control method for transmit antennas but also individual rate control method can be applied. The performance of the proposed technique is verified under a MIMO-OFDM (Orthogonal Frequency Division Multiplexing) system based on WLAN (Wireless Local Area Network), IEEE 802.11a. The results of the computer simulation show that a MIMO system with the proposed technique achieves higher throughput than one with a fixed transmission rate.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.46 no.3
• /
• pp.48-58
• /
• 2009

Geo-graphical routing protocols as GPSR are known to be very suitable and useful for vehicular ad-hoc networks. However, a corresponding node can include some stale neighbor nodes being out of a transmission range, and the stale nodes are pone to get a high priority to be a next relay node in the greedy mode. In addition, some useful redundant information can be eliminated during planarization in the recovery mode. This paper deals with a new recovery mode, the Greedy Border Superiority Routing(GBSR), along with an Adaptive Neighbor list Management(ANM) scheme. Each node can easily treat stale nodes on its neighbor list by means of comparing previous and current Position of a neighbor. When a node meets the local maximum, it makes use of a border superior graph to recover from it. This approach improve the packet delivery ratio while it decreases the time to recover from the local maximum. We evaluate the performance of the proposed methods using a network simulator. The results shown that the proposed protocol reveals much better performance than GPSR protocol. Please Put the of paper here.

• Journal of the Institute of Electronics Engineers of Korea CI
• /
• v.39 no.6
• /
• pp.53-67
• /
• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.